mirror of
https://github.com/librempeg/librempeg
synced 2024-11-21 16:44:05 +00:00
oss_audio: Split muxer and demuxer
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This commit is contained in:
Nidhi Makhijani
Diego Biurrun
parent
e0a2e60c0a
commit
d6e1d37100
@ -15,8 +15,8 @@ OBJS-$(CONFIG_BKTR_INDEV) += bktr.o
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OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o
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OBJS-$(CONFIG_FBDEV_INDEV) += fbdev.o
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OBJS-$(CONFIG_JACK_INDEV) += jack_audio.o timefilter.o
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OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o
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OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o
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OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o oss_audio_dec.o
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OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o oss_audio_enc.o
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OBJS-$(CONFIG_PULSE_INDEV) += pulse.o
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OBJS-$(CONFIG_SNDIO_INDEV) += sndio_common.o sndio_dec.o
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OBJS-$(CONFIG_SNDIO_OUTDEV) += sndio_common.o sndio_enc.o
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@ -20,45 +20,31 @@
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*/
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#include "config.h"
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#include <stdlib.h>
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#include <stdio.h>
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#include <stdint.h>
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#include <string.h>
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#include <errno.h>
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#if HAVE_SOUNDCARD_H
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#include <soundcard.h>
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#else
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#include <sys/soundcard.h>
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#endif
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#include <unistd.h>
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include "libavutil/internal.h"
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#include "libavutil/log.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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#include "libavcodec/avcodec.h"
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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#define AUDIO_BLOCK_SIZE 4096
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#include "oss_audio.h"
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typedef struct AudioData {
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AVClass *class;
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int fd;
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int sample_rate;
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int channels;
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int frame_size; /* in bytes ! */
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enum AVCodecID codec_id;
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unsigned int flip_left : 1;
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uint8_t buffer[AUDIO_BLOCK_SIZE];
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int buffer_ptr;
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} AudioData;
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static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
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int ff_oss_audio_open(AVFormatContext *s1, int is_output,
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const char *audio_device)
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{
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AudioData *s = s1->priv_data;
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OSSAudioData *s = s1->priv_data;
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int audio_fd;
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int tmp, err;
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char *flip = getenv("AUDIO_FLIP_LEFT");
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@ -80,7 +66,7 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
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if (!is_output)
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fcntl(audio_fd, F_SETFL, O_NONBLOCK);
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s->frame_size = AUDIO_BLOCK_SIZE;
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s->frame_size = OSS_AUDIO_BLOCK_SIZE;
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/* select format : favour native format */
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err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
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@ -143,183 +129,8 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
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return AVERROR(EIO);
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}
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static int audio_close(AudioData *s)
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int ff_oss_audio_close(OSSAudioData *s)
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{
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close(s->fd);
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return 0;
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}
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/* sound output support */
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static int audio_write_header(AVFormatContext *s1)
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{
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AudioData *s = s1->priv_data;
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AVStream *st;
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int ret;
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st = s1->streams[0];
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s->sample_rate = st->codec->sample_rate;
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s->channels = st->codec->channels;
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ret = audio_open(s1, 1, s1->filename);
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if (ret < 0) {
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return AVERROR(EIO);
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} else {
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return 0;
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}
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}
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static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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AudioData *s = s1->priv_data;
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int len, ret;
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int size= pkt->size;
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uint8_t *buf= pkt->data;
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while (size > 0) {
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len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
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memcpy(s->buffer + s->buffer_ptr, buf, len);
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s->buffer_ptr += len;
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if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
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for(;;) {
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ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
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if (ret > 0)
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break;
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if (ret < 0 && (errno != EAGAIN && errno != EINTR))
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return AVERROR(EIO);
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}
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s->buffer_ptr = 0;
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}
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buf += len;
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size -= len;
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}
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return 0;
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}
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static int audio_write_trailer(AVFormatContext *s1)
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{
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AudioData *s = s1->priv_data;
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audio_close(s);
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return 0;
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}
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/* grab support */
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static int audio_read_header(AVFormatContext *s1)
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{
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AudioData *s = s1->priv_data;
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AVStream *st;
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int ret;
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st = avformat_new_stream(s1, NULL);
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if (!st) {
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return AVERROR(ENOMEM);
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}
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ret = audio_open(s1, 0, s1->filename);
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if (ret < 0) {
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return AVERROR(EIO);
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}
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/* take real parameters */
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = s->codec_id;
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st->codec->sample_rate = s->sample_rate;
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st->codec->channels = s->channels;
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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return 0;
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}
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static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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AudioData *s = s1->priv_data;
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int ret, bdelay;
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int64_t cur_time;
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struct audio_buf_info abufi;
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if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
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return ret;
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ret = read(s->fd, pkt->data, pkt->size);
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if (ret <= 0){
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av_free_packet(pkt);
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pkt->size = 0;
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if (ret<0) return AVERROR(errno);
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else return AVERROR_EOF;
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}
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pkt->size = ret;
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/* compute pts of the start of the packet */
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cur_time = av_gettime();
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bdelay = ret;
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if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
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bdelay += abufi.bytes;
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}
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/* subtract time represented by the number of bytes in the audio fifo */
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cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
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/* convert to wanted units */
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pkt->pts = cur_time;
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if (s->flip_left && s->channels == 2) {
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int i;
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short *p = (short *) pkt->data;
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for (i = 0; i < ret; i += 4) {
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*p = ~*p;
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p += 2;
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}
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}
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return 0;
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}
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static int audio_read_close(AVFormatContext *s1)
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{
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AudioData *s = s1->priv_data;
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audio_close(s);
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return 0;
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}
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#if CONFIG_OSS_INDEV
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static const AVOption options[] = {
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{ "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass oss_demuxer_class = {
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.class_name = "OSS demuxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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AVInputFormat ff_oss_demuxer = {
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.name = "oss",
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.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
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.priv_data_size = sizeof(AudioData),
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.read_header = audio_read_header,
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.read_packet = audio_read_packet,
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.read_close = audio_read_close,
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.flags = AVFMT_NOFILE,
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.priv_class = &oss_demuxer_class,
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};
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#endif
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#if CONFIG_OSS_OUTDEV
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AVOutputFormat ff_oss_muxer = {
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.name = "oss",
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.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
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.priv_data_size = sizeof(AudioData),
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/* XXX: we make the assumption that the soundcard accepts this format */
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/* XXX: find better solution with "preinit" method, needed also in
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other formats */
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.audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
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.video_codec = AV_CODEC_ID_NONE,
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.write_header = audio_write_header,
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.write_packet = audio_write_packet,
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.write_trailer = audio_write_trailer,
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.flags = AVFMT_NOFILE,
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};
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#endif
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45
libavdevice/oss_audio.h
Normal file
45
libavdevice/oss_audio.h
Normal file
@ -0,0 +1,45 @@
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/*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVDEVICE_OSS_AUDIO_H
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#define AVDEVICE_OSS_AUDIO_H
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#include "libavcodec/avcodec.h"
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#include "libavformat/avformat.h"
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#define OSS_AUDIO_BLOCK_SIZE 4096
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typedef struct OSSAudioData {
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AVClass *class;
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int fd;
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int sample_rate;
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int channels;
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int frame_size; /* in bytes ! */
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enum AVCodecID codec_id;
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unsigned int flip_left : 1;
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uint8_t buffer[OSS_AUDIO_BLOCK_SIZE];
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int buffer_ptr;
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} OSSAudioData;
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int ff_oss_audio_open(AVFormatContext *s1, int is_output,
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const char *audio_device);
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int ff_oss_audio_close(OSSAudioData *s);
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#endif /* AVDEVICE_OSS_AUDIO_H */
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146
libavdevice/oss_audio_dec.c
Normal file
146
libavdevice/oss_audio_dec.c
Normal file
@ -0,0 +1,146 @@
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/*
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* Linux audio play interface
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* Copyright (c) 2000, 2001 Fabrice Bellard
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config.h"
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#include <stdint.h>
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#if HAVE_SOUNDCARD_H
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#include <soundcard.h>
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#else
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#include <sys/soundcard.h>
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#endif
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#include <unistd.h>
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include "libavutil/internal.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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#include "libavcodec/avcodec.h"
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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#include "oss_audio.h"
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static int audio_read_header(AVFormatContext *s1)
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{
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OSSAudioData *s = s1->priv_data;
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AVStream *st;
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int ret;
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st = avformat_new_stream(s1, NULL);
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if (!st) {
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return AVERROR(ENOMEM);
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}
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ret = ff_oss_audio_open(s1, 0, s1->filename);
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if (ret < 0) {
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return AVERROR(EIO);
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}
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/* take real parameters */
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = s->codec_id;
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st->codec->sample_rate = s->sample_rate;
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st->codec->channels = s->channels;
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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return 0;
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}
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static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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OSSAudioData *s = s1->priv_data;
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int ret, bdelay;
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int64_t cur_time;
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struct audio_buf_info abufi;
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if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
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return ret;
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ret = read(s->fd, pkt->data, pkt->size);
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if (ret <= 0){
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av_free_packet(pkt);
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pkt->size = 0;
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if (ret<0) return AVERROR(errno);
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else return AVERROR_EOF;
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}
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pkt->size = ret;
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/* compute pts of the start of the packet */
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cur_time = av_gettime();
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bdelay = ret;
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if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
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bdelay += abufi.bytes;
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}
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/* subtract time represented by the number of bytes in the audio fifo */
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cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
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/* convert to wanted units */
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pkt->pts = cur_time;
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if (s->flip_left && s->channels == 2) {
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int i;
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short *p = (short *) pkt->data;
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for (i = 0; i < ret; i += 4) {
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*p = ~*p;
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p += 2;
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}
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}
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return 0;
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}
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static int audio_read_close(AVFormatContext *s1)
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{
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OSSAudioData *s = s1->priv_data;
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ff_oss_audio_close(s);
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return 0;
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}
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static const AVOption options[] = {
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{ "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "channels", "", offsetof(OSSAudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass oss_demuxer_class = {
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.class_name = "OSS demuxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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AVInputFormat ff_oss_demuxer = {
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.name = "oss",
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.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
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.priv_data_size = sizeof(OSSAudioData),
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.read_header = audio_read_header,
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.read_packet = audio_read_packet,
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.read_close = audio_read_close,
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.flags = AVFMT_NOFILE,
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.priv_class = &oss_demuxer_class,
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};
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108
libavdevice/oss_audio_enc.c
Normal file
108
libavdevice/oss_audio_enc.c
Normal file
@ -0,0 +1,108 @@
|
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/*
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* Linux audio grab interface
|
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* Copyright (c) 2000, 2001 Fabrice Bellard
|
||||
*
|
||||
* This file is part of Libav.
|
||||
*
|
||||
* Libav is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Lesser General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2.1 of the License, or (at your option) any later version.
|
||||
*
|
||||
* Libav is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Lesser General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Lesser General Public
|
||||
* License along with Libav; if not, write to the Free Software
|
||||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
*/
|
||||
|
||||
#include "config.h"
|
||||
|
||||
#if HAVE_SOUNDCARD_H
|
||||
#include <soundcard.h>
|
||||
#else
|
||||
#include <sys/soundcard.h>
|
||||
#endif
|
||||
|
||||
#include <unistd.h>
|
||||
#include <fcntl.h>
|
||||
#include <sys/ioctl.h>
|
||||
|
||||
#include "libavutil/internal.h"
|
||||
|
||||
#include "libavcodec/avcodec.h"
|
||||
|
||||
#include "libavformat/avformat.h"
|
||||
#include "libavformat/internal.h"
|
||||
|
||||
#include "oss_audio.h"
|
||||
|
||||
static int audio_write_header(AVFormatContext *s1)
|
||||
{
|
||||
OSSAudioData *s = s1->priv_data;
|
||||
AVStream *st;
|
||||
int ret;
|
||||
|
||||
st = s1->streams[0];
|
||||
s->sample_rate = st->codec->sample_rate;
|
||||
s->channels = st->codec->channels;
|
||||
ret = ff_oss_audio_open(s1, 1, s1->filename);
|
||||
if (ret < 0) {
|
||||
return AVERROR(EIO);
|
||||
} else {
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
|
||||
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
|
||||
{
|
||||
OSSAudioData *s = s1->priv_data;
|
||||
int len, ret;
|
||||
int size= pkt->size;
|
||||
uint8_t *buf= pkt->data;
|
||||
|
||||
while (size > 0) {
|
||||
len = FFMIN(OSS_AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
|
||||
memcpy(s->buffer + s->buffer_ptr, buf, len);
|
||||
s->buffer_ptr += len;
|
||||
if (s->buffer_ptr >= OSS_AUDIO_BLOCK_SIZE) {
|
||||
for(;;) {
|
||||
ret = write(s->fd, s->buffer, OSS_AUDIO_BLOCK_SIZE);
|
||||
if (ret > 0)
|
||||
break;
|
||||
if (ret < 0 && (errno != EAGAIN && errno != EINTR))
|
||||
return AVERROR(EIO);
|
||||
}
|
||||
s->buffer_ptr = 0;
|
||||
}
|
||||
buf += len;
|
||||
size -= len;
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int audio_write_trailer(AVFormatContext *s1)
|
||||
{
|
||||
OSSAudioData *s = s1->priv_data;
|
||||
|
||||
ff_oss_audio_close(s);
|
||||
return 0;
|
||||
}
|
||||
|
||||
AVOutputFormat ff_oss_muxer = {
|
||||
.name = "oss",
|
||||
.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
|
||||
.priv_data_size = sizeof(OSSAudioData),
|
||||
/* XXX: we make the assumption that the soundcard accepts this format */
|
||||
/* XXX: find better solution with "preinit" method, needed also in
|
||||
other formats */
|
||||
.audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
|
||||
.video_codec = AV_CODEC_ID_NONE,
|
||||
.write_header = audio_write_header,
|
||||
.write_packet = audio_write_packet,
|
||||
.write_trailer = audio_write_trailer,
|
||||
.flags = AVFMT_NOFILE,
|
||||
};
|
||||
Loading…
Reference in New Issue
Block a user